similar to: getting bits per sample

Hi, I'm building an rtsp client/server that will support vorbis and I'm trying to determine the best way to play the vorbis content as the array is continuously being filled. Currently, I can download an ogg file via rtsp and write it out to a FILE and then play_file (from ogg123) it when it's done -- so the file is in the correct format. What would you recommend? -- Wayne Davis

Hi, I'm wondering if the plans for streaming support include on demand streams via rtsp -- not just a multicast of one stream like a radio station (icecast)?

Hello, I would like to implement the rtp specification. Is anyone else interested in helping with the design/code? Jack: i'm interested in your opinion on the difficulty of this since you worked on the rfc spec. Thanks. -- Wayne Davis - wfdavis@seas.upenn.edu - PGP Key Available --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

I'm trying to adapt decoder_example to to support more of an event driven model. I'm building an RTSP client/server where the client will have a function called everytime a new packet comes in. So, I want to queue up these packets to be played. My current attack is as follows (note this isn't very robust and I realized that): For first and second packet i grab all the header stuff

Hey guys. I think Vorbis is pretty cool, but since the current OggEnc only offers 44.1kHz, it limits what I wanted to use it for. So I've been using Lame to get 16kHz mono Vorbis files. I'm curious about whether Lame does Vorbis encoding the "right" way for non-44.1k stuff, or whether it just encodes as it would for 44.1k & changes the sample rate on the output, but I'm

--- Ingo Ralf Blum <ingoralfblum@gmx.de> wrote: > Hi, > > these two values can be stored in the stream info metadata block as > well as > in the header of each frame. This leads to some questions. > > - Are these values supposed to change during playback, e.g. first > comes a > frame with 2 channels, then with four and the next switches the bit > count >

I'm guessing here, but it seems that some encoders might only be able to process samples in 4-bit or 8-bit groups, maybe even 16-bit multiples, so this flag would allow the decoder to remove any padding that might have been necessary on the encoder side. For example, if someone were to encode 12-bit samples, but the encoder was forced to use 16-bit registers for calculations, then

2010/4/27 Brian Willoughby <brianw at sounds.wa.com> > i.e. I wouldn't be surprised if there are no FLAC files with this flag > set. > If you look for lossyWAV you'll find that it is actually being used -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac-dev/attachments/20100427/12821b2a/attachment.htm

On May 26, 2017, at 9:52 PM, Erik de Castro Lopo <mle+la at mega-nerd.com> wrote: > Mark H. David wrote: >> Notice the warning was given only for the WAVE file produced by the >> flac reference executable. > > I agree. The flac decoder should not generate WAV files which it > warns about when encoding. > >> Should I file a bug for this? > > No need.

Hi everyone, I'm trying to generate flac files with various bits-per-samples, using the 'flac' reference encoder and a raw audio file (I'm writing a decoder in Java and I need to create various test files). Creating 8, 16 or 24 bps works fine, but I didn't find how to generate other bps; the "--bps" option seems to accept only 8,16 or24. Any idea ? Thanks in

Dear list, i'm doing a bit of analisys on flac's source code and i've run into something i can't quite grasp. flac version 1.2.1 flaclib C stream_encoder.c function "process_subframes_" line 2999 ++++++++++++++++++++++ if(do_mid_side) { FLAC__ASSERT(encoder->protected_->channels == 2); for(channel = 0; channel < 2; channel++) {

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi, these two values can be stored in the stream info metadata block as well as in the header of each frame. This leads to some questions. - Are these values supposed to change during playback, e.g. first comes a frame with 2 channels, then with four and the next switches the bit count from 16 to 8 and so on? - There are methods to retrieve these values from the stream decoder. Unfortunately

Thank you for writing and pointing out that one can have WAVE files without getting the warning, and that your library does it. I found out Goldwave (Windows software) writes 24-bit WAVE files that do not get the warning. (Good for them!) However, the flac executable from the flac reference software apparently produces WAVE files that do get the warning. Example annotated log: Encode a 24-bit

Hello, I've tried encoding a FLAC file from a WAVE file saved with 24-bits-per-sample, using the flac executable. I get this warning: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 The code is currently here on line 373: src/flac/encode.c where the comment says let these slide with a warning since they're unambiguous I do not really understand the warning

On Wed, Mar 30, 2011 at 11:35 PM, Eli Friedman <eli.friedman at gmail.com> wrote: > On Wed, Mar 30, 2011 at 10:14 AM, Jimborean Alexandra > <xinfinity_a at yahoo.com> wrote: >> Hi, >> >> Is there any method to check if there is a path in the CFG from block M to >> block N, but M does not necessarily dominate block N? >> In other words, if N is

Hello, everybody. I am looking for same Ogg Vorbis file which have 8192 size frame or using floor0. According to the Ogg Vorbis spec, the maximum frame size is 8192. But I cannot find that size of frame in the Ogg Vorbis file. Also I cannot find Ogg Vorbis file using floor0. If you have, pleas open them. Or can you tell me where I can find them? Sincerely. Eunjik --- >8 ---- List

Hi, I'm a Pocket PC developer and have created a audio translator application on this platform. I currently make use of an MP3 decoder engine, but due to heavy licensing fees I'm thinking of switching over to a patent free format. Vorbis presents the best approach. I read that tremor decoder has been especially optimized for embedded devices. I'd like to make use of this and

https://bugzilla.samba.org/show_bug.cgi?id=1573 Summary: module revealed thru error msg even with "list = no" Product: rsync Version: 2.6.2 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: